Hi,
Guest
Sound issues: strange samplerates?
AT
Alain Totouom
Thu, Apr 7, 2016 3:03 PM
Hi Oliver
On 07/04/16 16:41, Oli Kah wrote:
Hi Alain,
thank you for answering :) Yes it seems that my VOIP router performs things it
should not do.
Currently I am looking for ways to ONLY include one specific codec: G711a.
But I don't know which defines should be set in config_site.h to achieve this.
Cheers,
Oliver
you can disable the audio codecs you don't need with the option
"--dis-codec=name".
Look at [1] for in depth-information.
Cheers,
Alain
Gesendet: Donnerstag, 07. April 2016 um 12:00 Uhr
Von: "Alain Totouom" alain.totouom@gmx.de
An: "pjsip list" pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
the callee (Fr!tzBox) did alter the supported codec list (offer/answer) between
his provisional (183/INVITE/cseq=6911) and his final response
(200/INVITE/cseq=6911) and did decline the callers request (UPDATE/cseq=6912) to
use the previously negotiated one (G722).
A more robust solution in a case like this, would i.E. imply caching the
answer-codec-list and updating the offer-codec-list for subsequent calls to the
same endpoint…
00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg INVITE/cseq=6911
(tdta003BBA90) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
[..]
00:58:12.347 pjsua_core.c .RX 755 bytes Response msg 183/INVITE/cseq=6911
(rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 9 0 8 96
[..]
00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv)
00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg 200/INVITE/cseq=6911
(rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 0 8 96
[..]
00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg UPDATE/cseq=6912
(tdta003D31C8) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 9 96
[..]
00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg 488/UPDATE/cseq=6912
(rdata003392C4) from UDP 192.168.1.1:5060:
Cheers,
Alain
On 07/04/16 11:14, Andreas Ahland wrote:
HI Oliver,
I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u.
Mit freundlichen Grüßen
Dr.-Ing.
Andreas Ahland
CTO
Technischer Leiter
Telefon : +49 251 6183-196
Telefax : +49 251 6183-197
Von: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Im Auftrag von Oli Kah
Gesendet: Donnerstag, 7. April 2016 01:13
An:pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Bill,
thanks for answering so promptly =)
I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?!
The three major problems currently for me are:
- it takes much too long to really establish a call (10 seconds are not normal)
- the sound received by the callee sounds strange, as if the sample rate is wrong
- the callee is NOT heard at all on the caller's side!
Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that?
Thank you so much!
Cheers,
Oliver
Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr
Von: "Bill Gardner" <billg@wavearts.com<mailto:billg@wavearts.com>>
An: "pjsip list" <pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>>
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there.
Bill
On 4/6/2016 4:26 PM, Oli Kah wrote:
Hi Bill, hi everyone,
thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h.
The result is exactly the same.
To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log.
There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear).
1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!!
2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected.
3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice!
The Python code of this mini app is attached to this email once more.
I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (seehttp://www.phoner.de/download_en.htm) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow.
Any ideas?
Thank you.
Cheers,
Oliver
Gesendet: Montag, 04. April 2016 um 20:49 Uhr
Von: "Bill Gardner"<billg@wavearts.com><billg@wavearts.com>
An:pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems.
Regards,
Bill
On 4/4/2016 2:37 PM, Oli Kah wrote:
Hmmm, no one?
Is there some sort of forum somewhere where to post things like these??
Thank you :)
Cheers,
Oli
Gesendet: Freitag, 01. April 2016 um 21:38 Uhr
Von: "Oli Kah"<mj_fn@web.de><mj_fn@web.de>
An:pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
Betreff: Sound issues: strange samplerates?
Hi there,
I am new to this list and want to say "Hello" to everyone listening :)
My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible!
What I did:
I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good.
These values were used for my config_site.h:
#define PJMEDIA_CONF_USE_SWITCH_BOARD 1
#define PJMEDIA_SOUND_BUFFER_COUNT 8
#define PJMEDIA_SND_DEFAULT_REC_LATENCY 50
#define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50
The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless!
During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream.
The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box").
The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long?
Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad.
The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge!
What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success.
Thank you!
Cheers,
Oliver
_______________________________________________
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
__________________________________________________________________________________
SELECTRIC Nachrichten-Systeme GmbH
Haferlandweg 18
48155 Muenster
Internet:
www.selectric.de
---
Sitz der Gesellschaft (Company Premises): Muenster | Registergericht (Register Court): Amtsgericht Muenster | HRB-Nr. 1264
Geschaeftsfuehrer (General Managers): Michael Heussner, Hendrik Pieper
_______________________________________________
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
--
1024D/A9F85A52 2000-01-18 Alain Totouom<totouom@gmx.de>
PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
_______________________________________________ Visit our blog:
http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
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1024D/A9F85A52 2000-01-18 Alain Totouom totouom@gmx.de
PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
Hi Oliver
On 07/04/16 16:41, Oli Kah wrote:
> Hi Alain,
> thank you for answering :) Yes it seems that my VOIP router performs things it
> should not do.
> Currently I am looking for ways to ONLY include one specific codec: G711a.
> But I don't know which defines should be set in config_site.h to achieve this.
> Cheers,
> Oliver
you can disable the audio codecs you don't need with the option
"--dis-codec=name".
Look at [1] for in depth-information.
Cheers,
Alain
[1] http://www.pjsip.org/pjsua.htm
> *Gesendet:* Donnerstag, 07. April 2016 um 12:00 Uhr
> *Von:* "Alain Totouom" <alain.totouom@gmx.de>
> *An:* "pjsip list" <pjsip@lists.pjsip.org>
> *Betreff:* Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> the callee (Fr!tzBox) did alter the supported codec list (offer/answer) between
> his provisional (183/INVITE/cseq=6911) and his final response
> (200/INVITE/cseq=6911) and did decline the callers request (UPDATE/cseq=6912) to
> use the previously negotiated one (G722).
> A more robust solution in a case like this, would i.E. imply caching the
> answer-codec-list and updating the offer-codec-list for subsequent calls to the
> same endpoint…
>
>
> 00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg INVITE/cseq=6911
> (tdta003BBA90) to UDP 192.168.1.1:5060:
> [..]
> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
> [..]
>
> 00:58:12.347 pjsua_core.c .RX 755 bytes Response msg 183/INVITE/cseq=6911
> (rdata003392C4) from UDP 192.168.1.1:5060:
> [..]
> m=audio 7086 RTP/AVP 9 0 8 96
> [..]
>
> 00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv)
> 00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg 200/INVITE/cseq=6911
> (rdata003392C4) from UDP 192.168.1.1:5060:
> [..]
> m=audio 7086 RTP/AVP 0 8 96
> [..]
>
> 00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg UPDATE/cseq=6912
> (tdta003D31C8) to UDP 192.168.1.1:5060:
> [..]
> m=audio 4000 RTP/AVP 9 96
> [..]
>
> 00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg 488/UPDATE/cseq=6912
> (rdata003392C4) from UDP 192.168.1.1:5060:
>
> Cheers,
> Alain
> On 07/04/16 11:14, Andreas Ahland wrote:
>
> HI Oliver,
>
> I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u.
>
> Mit freundlichen Grüßen
>
> Dr.-Ing.
> Andreas Ahland
> CTO
> Technischer Leiter
> Telefon : +49 251 6183-196
> Telefax : +49 251 6183-197
>
>
>
>
> Von: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Im Auftrag von Oli Kah
> Gesendet: Donnerstag, 7. April 2016 01:13
> An:pjsip@lists.pjsip.org
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
>
> Hi Bill,
>
> thanks for answering so promptly =)
> I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?!
>
> The three major problems currently for me are:
> - it takes much too long to really establish a call (10 seconds are not normal)
> - the sound received by the callee sounds strange, as if the sample rate is wrong
> - the callee is NOT heard at all on the caller's side!
>
> Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that?
>
> Thank you so much!
>
> Cheers,
> Oliver
>
> Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr
> Von: "Bill Gardner" <billg@wavearts.com<mailto:billg@wavearts.com>>
> An: "pjsip list" <pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>>
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there.
>
> Bill
>
> On 4/6/2016 4:26 PM, Oli Kah wrote:
> Hi Bill, hi everyone,
>
> thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h.
> The result is exactly the same.
>
> To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log.
>
> There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear).
>
> 1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!!
> 2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected.
> 3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice!
>
> The Python code of this mini app is attached to this email once more.
>
> I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (seehttp://www.phoner.de/download_en.htm) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow.
>
> Any ideas?
>
> Thank you.
>
> Cheers,
> Oliver
>
> Gesendet: Montag, 04. April 2016 um 20:49 Uhr
> Von: "Bill Gardner"<billg@wavearts.com><billg@wavearts.com>
> An:pjsip@lists.pjsip.org
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems.
>
> Regards,
>
> Bill
>
> On 4/4/2016 2:37 PM, Oli Kah wrote:
> Hmmm, no one?
>
> Is there some sort of forum somewhere where to post things like these??
>
> Thank you :)
>
> Cheers,
> Oli
>
> Gesendet: Freitag, 01. April 2016 um 21:38 Uhr
> Von: "Oli Kah"<mj_fn@web.de><mj_fn@web.de>
> An:pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
> Betreff: Sound issues: strange samplerates?
> Hi there,
>
>
> I am new to this list and want to say "Hello" to everyone listening :)
>
> My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible!
>
> What I did:
>
> I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good.
> These values were used for my config_site.h:
>
> #define PJMEDIA_CONF_USE_SWITCH_BOARD 1
> #define PJMEDIA_SOUND_BUFFER_COUNT 8
> #define PJMEDIA_SND_DEFAULT_REC_LATENCY 50
> #define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50
>
> The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless!
>
> During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream.
>
> The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box").
>
> The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long?
> Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad.
>
> The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge!
>
>
> What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success.
>
>
> Thank you!
>
> Cheers,
> Oliver
>
>
>
> _______________________________________________
>
> Visit our blog:http://blog.pjsip.org
>
>
>
> pjsip mailing list
>
> pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
>
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
> _______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
> _______________________________________________
>
> Visit our blog:http://blog.pjsip.org
>
>
>
> pjsip mailing list
>
> pjsip@lists.pjsip.org
>
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
> _______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> __________________________________________________________________________________
> SELECTRIC Nachrichten-Systeme GmbH
> Haferlandweg 18
> 48155 Muenster
>
> Internet:
> www.selectric.de
>
> ---
> Sitz der Gesellschaft (Company Premises): Muenster | Registergericht (Register Court): Amtsgericht Muenster | HRB-Nr. 1264
> Geschaeftsfuehrer (General Managers): Michael Heussner, Hendrik Pieper
>
> _______________________________________________
> Visit our blog:http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
> --
> 1024D/A9F85A52 2000-01-18 Alain Totouom<totouom@gmx.de>
> PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
>
> _______________________________________________ Visit our blog:
> http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
> --
> 1024D/A9F85A52 2000-01-18 Alain Totouom <totouom@gmx.de>
> PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
BG
Bill Gardner
Thu, Apr 7, 2016 3:19 PM
Hi Oliver,
I agree with the other responders. Disable everything except G711 (PCMU)
using the --dis-codec option and it will probably work better. You may
also want to wait for the REGISTER to complete before starting the INVITE.
Bill
On 4/7/2016 10:41 AM, Oli Kah wrote:
Hi Alain,
thank you for answering :) Yes it seems that my VOIP router performs
things it should not do.
Currently I am looking for ways to ONLY include one specific codec: G711a.
But I don't know which defines should be set in config_site.h to
achieve this.
Cheers,
Oliver
Gesendet: Donnerstag, 07. April 2016 um 12:00 Uhr
Von: "Alain Totouom" alain.totouom@gmx.de
An: "pjsip list" pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
the callee (Fr!tzBox) did alter the supported codec list
(offer/answer) between his provisional (183/INVITE/cseq=6911) and his
final response (200/INVITE/cseq=6911) and did decline the callers
request (UPDATE/cseq=6912) to use the previously negotiated one (G722).
A more robust solution in a case like this, would i.E. imply caching
the answer-codec-list and updating the offer-codec-list for subsequent
calls to the same endpoint…
00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg
INVITE/cseq=6911 (tdta003BBA90) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
[..]
00:58:12.347 pjsua_core.c .RX 755 bytes Response msg
183/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 9 0 8 96
[..]
00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722
(sendrecv)
00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg
200/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 0 8 96
[..]
00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg
UPDATE/cseq=6912 (tdta003D31C8) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 9 96
[..]
00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg
488/UPDATE/cseq=6912 (rdata003392C4) from UDP 192.168.1.1:5060:
Cheers,
Alain
On 07/04/16 11:14, Andreas Ahland wrote:
HI Oliver,
I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u.
Mit freundlichen Grüßen
Dr.-Ing.
Andreas Ahland
CTO
Technischer Leiter
Telefon : +49 251 6183-196
Telefax : +49 251 6183-197
Von: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Im Auftrag von Oli Kah
Gesendet: Donnerstag, 7. April 2016 01:13
An:pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Bill,
thanks for answering so promptly =)
I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?!
The three major problems currently for me are:
- it takes much too long to really establish a call (10 seconds are not normal)
- the sound received by the callee sounds strange, as if the sample rate is wrong
- the callee is NOT heard at all on the caller's side!
Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that?
Thank you so much!
Cheers,
Oliver
Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr
Von: "Bill Gardner" <billg@wavearts.com<mailto:billg@wavearts.com>>
An: "pjsip list" <pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>>
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there.
Bill
On 4/6/2016 4:26 PM, Oli Kah wrote:
Hi Bill, hi everyone,
thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h.
The result is exactly the same.
To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log.
There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear).
1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!!
2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected.
3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice!
The Python code of this mini app is attached to this email once more.
I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (seehttp://www.phoner.de/download_en.htm) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow.
Any ideas?
Thank you.
Cheers,
Oliver
Gesendet: Montag, 04. April 2016 um 20:49 Uhr
Von: "Bill Gardner"<billg@wavearts.com><billg@wavearts.com>
An:pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems.
Regards,
Bill
On 4/4/2016 2:37 PM, Oli Kah wrote:
Hmmm, no one?
Is there some sort of forum somewhere where to post things like these??
Thank you :)
Cheers,
Oli
Gesendet: Freitag, 01. April 2016 um 21:38 Uhr
Von: "Oli Kah"<mj_fn@web.de><mj_fn@web.de>
An:pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
Betreff: Sound issues: strange samplerates?
Hi there,
I am new to this list and want to say "Hello" to everyone listening :)
My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible!
What I did:
I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good.
These values were used for my config_site.h:
#define PJMEDIA_CONF_USE_SWITCH_BOARD 1
#define PJMEDIA_SOUND_BUFFER_COUNT 8
#define PJMEDIA_SND_DEFAULT_REC_LATENCY 50
#define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50
The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless!
During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream.
The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box").
The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long?
Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad.
The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge!
What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success.
Thank you!
Cheers,
Oliver
_______________________________________________
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pjsip mailing list
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pjsip mailing list
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_______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
__________________________________________________________________________________
SELECTRIC Nachrichten-Systeme GmbH
Haferlandweg 18
48155 Muenster
Internet:
www.selectric.de
---
Sitz der Gesellschaft (Company Premises): Muenster | Registergericht (Register Court): Amtsgericht Muenster | HRB-Nr. 1264
Geschaeftsfuehrer (General Managers): Michael Heussner, Hendrik Pieper
_______________________________________________
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--
1024D/A9F85A52 2000-01-18 Alain Totouom<totouom@gmx.de>
PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
_______________________________________________ Visit our blog:
http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org
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pjsip mailing list
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Hi Oliver,
I agree with the other responders. Disable everything except G711 (PCMU)
using the --dis-codec option and it will probably work better. You may
also want to wait for the REGISTER to complete before starting the INVITE.
Bill
On 4/7/2016 10:41 AM, Oli Kah wrote:
> Hi Alain,
> thank you for answering :) Yes it seems that my VOIP router performs
> things it should not do.
> Currently I am looking for ways to ONLY include one specific codec: G711a.
> But I don't know which defines should be set in config_site.h to
> achieve this.
> Cheers,
> Oliver
> *Gesendet:* Donnerstag, 07. April 2016 um 12:00 Uhr
> *Von:* "Alain Totouom" <alain.totouom@gmx.de>
> *An:* "pjsip list" <pjsip@lists.pjsip.org>
> *Betreff:* Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> the callee (Fr!tzBox) did alter the supported codec list
> (offer/answer) between his provisional (183/INVITE/cseq=6911) and his
> final response (200/INVITE/cseq=6911) and did decline the callers
> request (UPDATE/cseq=6912) to use the previously negotiated one (G722).
> A more robust solution in a case like this, would i.E. imply caching
> the answer-codec-list and updating the offer-codec-list for subsequent
> calls to the same endpoint…
>
>
> 00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg
> INVITE/cseq=6911 (tdta003BBA90) to UDP 192.168.1.1:5060:
> [..]
> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
> [..]
>
> 00:58:12.347 pjsua_core.c .RX 755 bytes Response msg
> 183/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
> [..]
> m=audio 7086 RTP/AVP 9 0 8 96
> [..]
>
> 00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722
> (sendrecv)
> 00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg
> 200/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
> [..]
> m=audio 7086 RTP/AVP 0 8 96
> [..]
>
> 00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg
> UPDATE/cseq=6912 (tdta003D31C8) to UDP 192.168.1.1:5060:
> [..]
> m=audio 4000 RTP/AVP 9 96
> [..]
>
> 00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg
> 488/UPDATE/cseq=6912 (rdata003392C4) from UDP 192.168.1.1:5060:
>
> Cheers,
> Alain
> On 07/04/16 11:14, Andreas Ahland wrote:
>
> HI Oliver,
>
> I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u.
>
> Mit freundlichen Grüßen
>
> Dr.-Ing.
> Andreas Ahland
> CTO
> Technischer Leiter
> Telefon : +49 251 6183-196
> Telefax : +49 251 6183-197
>
>
>
>
> Von: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Im Auftrag von Oli Kah
> Gesendet: Donnerstag, 7. April 2016 01:13
> An:pjsip@lists.pjsip.org
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
>
> Hi Bill,
>
> thanks for answering so promptly =)
> I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?!
>
> The three major problems currently for me are:
> - it takes much too long to really establish a call (10 seconds are not normal)
> - the sound received by the callee sounds strange, as if the sample rate is wrong
> - the callee is NOT heard at all on the caller's side!
>
> Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that?
>
> Thank you so much!
>
> Cheers,
> Oliver
>
> Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr
> Von: "Bill Gardner" <billg@wavearts.com<mailto:billg@wavearts.com>>
> An: "pjsip list" <pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>>
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there.
>
> Bill
>
> On 4/6/2016 4:26 PM, Oli Kah wrote:
> Hi Bill, hi everyone,
>
> thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h.
> The result is exactly the same.
>
> To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log.
>
> There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear).
>
> 1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!!
> 2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected.
> 3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice!
>
> The Python code of this mini app is attached to this email once more.
>
> I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (seehttp://www.phoner.de/download_en.htm) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow.
>
> Any ideas?
>
> Thank you.
>
> Cheers,
> Oliver
>
> Gesendet: Montag, 04. April 2016 um 20:49 Uhr
> Von: "Bill Gardner"<billg@wavearts.com><billg@wavearts.com>
> An:pjsip@lists.pjsip.org
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems.
>
> Regards,
>
> Bill
>
> On 4/4/2016 2:37 PM, Oli Kah wrote:
> Hmmm, no one?
>
> Is there some sort of forum somewhere where to post things like these??
>
> Thank you :)
>
> Cheers,
> Oli
>
> Gesendet: Freitag, 01. April 2016 um 21:38 Uhr
> Von: "Oli Kah"<mj_fn@web.de><mj_fn@web.de>
> An:pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
> Betreff: Sound issues: strange samplerates?
> Hi there,
>
>
> I am new to this list and want to say "Hello" to everyone listening :)
>
> My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible!
>
> What I did:
>
> I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good.
> These values were used for my config_site.h:
>
> #define PJMEDIA_CONF_USE_SWITCH_BOARD 1
> #define PJMEDIA_SOUND_BUFFER_COUNT 8
> #define PJMEDIA_SND_DEFAULT_REC_LATENCY 50
> #define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50
>
> The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless!
>
> During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream.
>
> The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box").
>
> The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long?
> Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad.
>
> The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge!
>
>
> What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success.
>
>
> Thank you!
>
> Cheers,
> Oliver
>
>
>
> _______________________________________________
>
> Visit our blog:http://blog.pjsip.org
>
>
>
> pjsip mailing list
>
> pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
>
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
> _______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
> _______________________________________________
>
> Visit our blog:http://blog.pjsip.org
>
>
>
> pjsip mailing list
>
> pjsip@lists.pjsip.org
>
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
> _______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> __________________________________________________________________________________
> SELECTRIC Nachrichten-Systeme GmbH
> Haferlandweg 18
> 48155 Muenster
>
> Internet:
> www.selectric.de
>
> ---
> Sitz der Gesellschaft (Company Premises): Muenster | Registergericht (Register Court): Amtsgericht Muenster | HRB-Nr. 1264
> Geschaeftsfuehrer (General Managers): Michael Heussner, Hendrik Pieper
>
> _______________________________________________
> Visit our blog:http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
> --
> 1024D/A9F85A52 2000-01-18 Alain Totouom<totouom@gmx.de>
> PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
>
> _______________________________________________ Visit our blog:
> http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip@lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
OK
Oli Kah
Thu, Apr 7, 2016 3:23 PM
Hi Alain,
thanks for responding again however I do not use the command line tool but I am working entirely in Python with my SIP phone client.
What I potentially can do however is recompile the pjsua library in a way that it does not include any "unwanted" codec.
With these settings in config_site.h:
#define PJMEDIA_HAS_G711_CODEC 1
#define PJMEDIA_HAS_FFMPEG 0
#define PJMEDIA_HAS_L16_CODEC 0
#define PJMEDIA_HAS_G722_CODEC 0
#define PJMEDIA_HAS_GSM_CODEC 0
I was able to eliminate most of the codecs - but not all of them!
Here is a test output (produced by Python code) of the enumerated codecs that remain this way:
HAVE CODEC: speex/16000/1
HAVE CODEC: speex/8000/1
HAVE CODEC: speex/32000/1
HAVE CODEC: iLBC/8000/1
HAVE CODEC: PCMU/8000/1
HAVE CODEC: PCMA/8000/1
So following Andreas' answer I only want to include G711a but it seems like G711u is still included as well as speex and iLBC.
I do not know much about SIP audio codecs so for me the questions remains how to get a decent result removing as many codecs as possible without loosing too many other features (such as the integrated echo canceller).
By now I can talk normally without any monster voice and the connection is established much faster but still not always promptly. However the major inital problems are mostly gone by removing some of the codecs. But there is a new surprise happening now as the media gets active TWICE which is a bit strange so I hope to get rid of the other codecs as well to see if that helps.
Thanks for your help in any case =)
Cheers,
Oliver
Gesendet: Donnerstag, 07. April 2016 um 17:03 Uhr
Von: "Alain Totouom" <alain.totouom@gmx.de>
An: "pjsip list" <pjsip@lists.pjsip.org>
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver
On 07/04/16 16:41, Oli Kah wrote: > ```
Hi Alain,
thank you for answering :) Yes it seems that my VOIP router performs things it
should not do.
Currently I am looking for ways to ONLY include one specific codec: G711a.
But I don't know which defines should be set in config_site.h to achieve this.
Cheers,
Oliver
you can disable the audio codecs you don't need with the option "--dis-codec=name".
Look at [1] for in depth-information.
Cheers,
Alain
[1] http://www.pjsip.org/pjsua.htm
*Gesendet:* Donnerstag, 07. April 2016 um 12:00 Uhr *Von:* "Alain Totouom" <a class="moz-txt-link-rfc2396E" href="alain.totouom@gmx.de" target="_parent"><alain.totouom@gmx.de></a> *An:* "pjsip list" <a class="moz-txt-link-rfc2396E" href="pjsip@lists.pjsip.org" target="_parent"><pjsip@lists.pjsip.org></a> *Betreff:* Re: [pjsip] Fw: Sound issues: strange samplerates? Hi Oliver, the callee (Fr!tzBox) did alter the supported codec list (offer/answer) between his provisional (183/INVITE/cseq=6911) and his final response (200/INVITE/cseq=6911) and did decline the callers request (UPDATE/cseq=6912) to use the previously negotiated one (G722). A more robust solution in a case like this, would i.E. imply caching the answer-codec-list and updating the offer-codec-list for subsequent calls to the same endpoint… 00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg INVITE/cseq=6911 (tdta003BBA90) to UDP 192.168.1.1:5060: [..] m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 [..] 00:58:12.347 pjsua_core.c .RX 755 bytes Response msg 183/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060: [..] m=audio 7086 RTP/AVP 9 0 8 96 [..] 00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv) 00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg 200/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060: [..] m=audio 7086 RTP/AVP 0 8 96 [..] 00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg UPDATE/cseq=6912 (tdta003D31C8) to UDP 192.168.1.1:5060: [..] m=audio 4000 RTP/AVP 9 96 [..] 00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg 488/UPDATE/cseq=6912 (rdata003392C4) from UDP 192.168.1.1:5060: Cheers, Alain On 07/04/16 11:14, Andreas Ahland wrote: HI Oliver, I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u. Mit freundlichen Grüßen Dr.-Ing. Andreas Ahland CTO Technischer Leiter Telefon : +49 251 6183-196 Telefax : +49 251 6183-197 Von: pjsip [<a class="moz-txt-link-freetext" href="pjsip-bounces@lists.pjsip.org" target="_parent">mailto:pjsip-bounces@lists.pjsip.org</a>] Im Auftrag von Oli Kah Gesendet: Donnerstag, 7. April 2016 01:13 <a class="moz-txt-link-abbreviated" href="An:pjsip@lists.pjsip.org" target="_parent">An:pjsip@lists.pjsip.org</a> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates? Hi Bill, thanks for answering so promptly =) I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?! The three major problems currently for me are: - it takes much too long to really establish a call (10 seconds are not normal) - the sound received by the callee sounds strange, as if the sample rate is wrong - the callee is NOT heard at all on the caller's side! Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that? Thank you so much! Cheers, Oliver Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr Von: "Bill Gardner" <<a class="moz-txt-link-abbreviated" href="billg@wavearts.com" target="_parent">billg@wavearts.com</a><a class="moz-txt-link-rfc2396E" href="billg@wavearts.com" target="_parent"><mailto:billg@wavearts.com></a>> An: "pjsip list" <<a class="moz-txt-link-abbreviated" href="pjsip@lists.pjsip.org" target="_parent">pjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E" href="pjsip@lists.pjsip.org" target="_parent"><mailto:pjsip@lists.pjsip.org></a>> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates? Hi Oliver, Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there. Bill On 4/6/2016 4:26 PM, Oli Kah wrote: Hi Bill, hi everyone, thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h. The result is exactly the same. To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log. There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear). 1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!! 2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected. 3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice! The Python code of this mini app is attached to this email once more. I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (seehttp://www.phoner.de/download_en.htm) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow. Any ideas? Thank you. Cheers, Oliver Gesendet: Montag, 04. April 2016 um 20:49 Uhr Von: "Bill Gardner"<a class="moz-txt-link-rfc2396E" href="billg@wavearts.com" target="_parent"><billg@wavearts.com></a><a class="moz-txt-link-rfc2396E" href="billg@wavearts.com" target="_parent"><billg@wavearts.com></a> <a class="moz-txt-link-abbreviated" href="An:pjsip@lists.pjsip.org" target="_parent">An:pjsip@lists.pjsip.org</a> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates? Hi Oliver, I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems. Regards, Bill On 4/4/2016 2:37 PM, Oli Kah wrote: Hmmm, no one? Is there some sort of forum somewhere where to post things like these?? Thank you :) Cheers, Oli Gesendet: Freitag, 01. April 2016 um 21:38 Uhr Von: "Oli Kah"<a class="moz-txt-link-rfc2396E" href="mj_fn@web.de" target="_parent"><mj_fn@web.de></a><a class="moz-txt-link-rfc2396E" href="mj_fn@web.de" target="_parent"><mj_fn@web.de></a> <a class="moz-txt-link-abbreviated" href="An:pjsip@lists.pjsip.org" target="_parent">An:pjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E" href="pjsip@lists.pjsip.org" target="_parent"><mailto:pjsip@lists.pjsip.org></a> Betreff: Sound issues: strange samplerates? Hi there, I am new to this list and want to say "Hello" to everyone listening :) My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible! What I did: I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good. These values were used for my config_site.h: #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 #define PJMEDIA_SOUND_BUFFER_COUNT 8 #define PJMEDIA_SND_DEFAULT_REC_LATENCY 50 #define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50 The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless! During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream. The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box"). The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long? Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad. The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge! What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success. Thank you! Cheers, Oliver _______________________________________________ Visit our blog:<a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated" href="pjsip@lists.pjsip.org" target="_parent">pjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E" href="pjsip@lists.pjsip.org" target="_parent"><mailto:pjsip@lists.pjsip.org></a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> _______________________________________________ Visit our blog:<a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing <a class="moz-txt-link-abbreviated" href="listpjsip@lists.pjsip.org" target="_parent">listpjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> _______________________________________________ Visit our blog:<a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated" href="pjsip@lists.pjsip.org" target="_parent">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> _______________________________________________ Visit our blog:<a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing <a class="moz-txt-link-abbreviated" href="listpjsip@lists.pjsip.org" target="_parent">listpjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E" href="pjsip@lists.pjsip.org" target="_parent"><mailto:pjsip@lists.pjsip.org></a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> __________________________________________________________________________________ SELECTRIC Nachrichten-Systeme GmbH Haferlandweg 18 48155 Muenster Internet: <a class="moz-txt-link-abbreviated" href="http://www.selectric.de" target="_blank">www.selectric.de</a> --- Sitz der Gesellschaft (Company Premises): Muenster | Registergericht (Register Court): Amtsgericht Muenster | HRB-Nr. 1264 Geschaeftsfuehrer (General Managers): Michael Heussner, Hendrik Pieper _______________________________________________ Visit our blog:<a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated" href="pjsip@lists.pjsip.org" target="_parent">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> -- 1024D/A9F85A52 2000-01-18 Alain Totouom<a class="moz-txt-link-rfc2396E" href="totouom@gmx.de" target="_parent"><totouom@gmx.de></a> PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52 _______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated" href="pjsip@lists.pjsip.org" target="_parent">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a>_______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated" href="pjsip@lists.pjsip.org" target="_parent">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a><pre class="moz-signature">-- 1024D/A9F85A52 2000-01-18 Alain Totouom <a class="moz-txt-link-rfc2396E" href="totouom@gmx.de" target="_parent"><totouom@gmx.de></a> PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
_______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
OK
Oli Kah
Thu, Apr 7, 2016 3:27 PM
Hi Bill,
thanks for your support :)
Yes, it seems like the SIP support of my router produced these issues and I agree also to remove all codecs that lead to this behavior. However I am not working with the command line version of pjsip but only using the Python bindings.
So I have to find a way to eliminate the "unwanted" codecs when compiling the Python library somehow.
Thanks again!
Cheers,
Oliver
Gesendet: Donnerstag, 07. April 2016 um 17:19 Uhr
Von: "Bill Gardner" <billg@wavearts.com>
An: pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
I agree with the other responders. Disable everything except G711 (PCMU) using the --dis-codec option and it will probably work better. You may also want to wait for the REGISTER to complete before starting the INVITE.
Bill
On 4/7/2016 10:41 AM, Oli Kah wrote:
Hi Alain,
thank you for answering :) Yes it seems that my VOIP router performs things it should not do.
Currently I am looking for ways to ONLY include one specific codec: G711a.
But I don't know which defines should be set in config_site.h to achieve this.
Cheers,
Oliver
Gesendet: Donnerstag, 07. April 2016 um 12:00 Uhr
Von: "Alain Totouom" <alain.totouom@gmx.de>
An: "pjsip list" <pjsip@lists.pjsip.org>
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?Hi Oliver,
the callee (Fr!tzBox) did alter the supported codec list (offer/answer) between his provisional (183/INVITE/cseq=6911) and his final response (200/INVITE/cseq=6911) and did decline the callers request (UPDATE/cseq=6912) to use the previously negotiated one (G722).
A more robust solution in a case like this, would i.E. imply caching the answer-codec-list and updating the offer-codec-list for subsequent calls to the same endpoint…00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg INVITE/cseq=6911 (tdta003BBA90) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
[..]00:58:12.347 pjsua_core.c .RX 755 bytes Response msg 183/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 9 0 8 96
[..]00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv)
00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg 200/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 0 8 96
[..]00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg UPDATE/cseq=6912 (tdta003D31C8) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 9 96
[..]00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg 488/UPDATE/cseq=6912 (rdata003392C4) from UDP 192.168.1.1:5060:
Cheers,
Alain
On 07/04/16 11:14, Andreas Ahland wrote:HI Oliver, I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u. Mit freundlichen Grüßen Dr.-Ing. Andreas Ahland CTO Technischer Leiter Telefon : +49 251 6183-196 Telefax : +49 251 6183-197 Von: pjsip [<a class="moz-txt-link-freetext">mailto:pjsip-bounces@lists.pjsip.org</a>] Im Auftrag von Oli Kah Gesendet: Donnerstag, 7. April 2016 01:13 An: <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates? Hi Bill, thanks for answering so promptly =) I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?! The three major problems currently for me are: - it takes much too long to really establish a call (10 seconds are not normal) - the sound received by the callee sounds strange, as if the sample rate is wrong - the callee is NOT heard at all on the caller's side! Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that? Thank you so much! Cheers, Oliver Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr Von: "Bill Gardner" <<a class="moz-txt-link-abbreviated">billg@wavearts.com</a><a class="moz-txt-link-rfc2396E"><mailto:billg@wavearts.com></a>> An: "pjsip list" <<a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E"><mailto:pjsip@lists.pjsip.org></a>> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates? Hi Oliver, Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there. Bill On 4/6/2016 4:26 PM, Oli Kah wrote: Hi Bill, hi everyone, thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h. The result is exactly the same. To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log. There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear). 1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!! 2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected. 3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice! The Python code of this mini app is attached to this email once more. I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (see <a class="moz-txt-link-freetext" href="http://www.phoner.de/download_en.htm" target="_blank">http://www.phoner.de/download_en.htm</a>) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow. Any ideas? Thank you. Cheers, Oliver Gesendet: Montag, 04. April 2016 um 20:49 Uhr Von: "Bill Gardner" <a class="moz-txt-link-rfc2396E"><billg@wavearts.com></a><a class="moz-txt-link-rfc2396E"><billg@wavearts.com></a> An: <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates? Hi Oliver, I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems. Regards, Bill On 4/4/2016 2:37 PM, Oli Kah wrote: Hmmm, no one? Is there some sort of forum somewhere where to post things like these?? Thank you :) Cheers, Oli Gesendet: Freitag, 01. April 2016 um 21:38 Uhr Von: "Oli Kah" <a class="moz-txt-link-rfc2396E"><mj_fn@web.de></a><a class="moz-txt-link-rfc2396E"><mj_fn@web.de></a> An: <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E"><mailto:pjsip@lists.pjsip.org></a> Betreff: Sound issues: strange samplerates? Hi there, I am new to this list and want to say "Hello" to everyone listening :) My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible! What I did: I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good. These values were used for my config_site.h: #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 #define PJMEDIA_SOUND_BUFFER_COUNT 8 #define PJMEDIA_SND_DEFAULT_REC_LATENCY 50 #define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50 The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless! During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream. The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box"). The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long? Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad. The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge! What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success. Thank you! Cheers, Oliver _______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E"><mailto:pjsip@lists.pjsip.org></a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> _______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> _______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> _______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a><a class="moz-txt-link-rfc2396E"><mailto:pjsip@lists.pjsip.org></a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a> __________________________________________________________________________________ SELECTRIC Nachrichten-Systeme GmbH Haferlandweg 18 48155 Muenster Internet: <a class="moz-txt-link-abbreviated" href="http://www.selectric.de" target="_blank">www.selectric.de</a> --- Sitz der Gesellschaft (Company Premises): Muenster | Registergericht (Register Court): Amtsgericht Muenster | HRB-Nr. 1264 Geschaeftsfuehrer (General Managers): Michael Heussner, Hendrik Pieper_______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a><pre class="moz-signature">-- 1024D/A9F85A52 2000-01-18 Alain Totouom <a class="moz-txt-link-rfc2396E"><totouom@gmx.de></a> PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52_______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________ Visit our blog: <a class="moz-txt-link-freetext" href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a> pjsip mailing list <a class="moz-txt-link-abbreviated" href="pjsip@lists.pjsip.org" target="_parent">pjsip@lists.pjsip.org</a> <a class="moz-txt-link-freetext" href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a>
_______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
BG
Bill Gardner
Thu, Apr 7, 2016 3:48 PM
There are config options for each code, so you can get rid of them that
way (look in the code for each codec to see the option), but you'll end
up with both PCMU and PCMA. That's probably OK.
Bill
On 4/7/2016 11:27 AM, Oli Kah wrote:
Hi Bill,
thanks for your support :)
Yes, it seems like the SIP support of my router produced these issues
and I agree also to remove all codecs that lead to this behavior.
However I am not working with the command line version of pjsip but
only using the Python bindings.
So I have to find a way to eliminate the "unwanted" codecs when
compiling the Python library somehow.
Thanks again!
Cheers,
Oliver
Gesendet: Donnerstag, 07. April 2016 um 17:19 Uhr
Von: "Bill Gardner" billg@wavearts.com
An: pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
I agree with the other responders. Disable everything except G711
(PCMU) using the --dis-codec option and it will probably work better.
You may also want to wait for the REGISTER to complete before starting
the INVITE.
Bill
On 4/7/2016 10:41 AM, Oli Kah wrote:
Hi Alain,
thank you for answering :) Yes it seems that my VOIP router
performs things it should not do.
Currently I am looking for ways to ONLY include one specific
codec: G711a.
But I don't know which defines should be set in config_site.h to
achieve this.
Cheers,
Oliver
*Gesendet:* Donnerstag, 07. April 2016 um 12:00 Uhr
*Von:* "Alain Totouom" <alain.totouom@gmx.de>
*An:* "pjsip list" <pjsip@lists.pjsip.org>
*Betreff:* Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
the callee (Fr!tzBox) did alter the supported codec list
(offer/answer) between his provisional (183/INVITE/cseq=6911) and
his final response (200/INVITE/cseq=6911) and did decline the
callers request (UPDATE/cseq=6912) to use the previously
negotiated one (G722).
A more robust solution in a case like this, would i.E. imply
caching the answer-codec-list and updating the offer-codec-list
for subsequent calls to the same endpoint…
00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg
INVITE/cseq=6911 (tdta003BBA90) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
[..]
00:58:12.347 pjsua_core.c .RX 755 bytes Response msg
183/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 9 0 8 96
[..]
00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722
(sendrecv)
00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg
200/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
[..]
m=audio 7086 RTP/AVP 0 8 96
[..]
00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg
UPDATE/cseq=6912 (tdta003D31C8) to UDP 192.168.1.1:5060:
[..]
m=audio 4000 RTP/AVP 9 96
[..]
00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg
488/UPDATE/cseq=6912 (rdata003392C4) from UDP 192.168.1.1:5060:
Cheers,
Alain
On 07/04/16 11:14, Andreas Ahland wrote:
HI Oliver,
I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u.
Mit freundlichen Grüßen
Dr.-Ing.
Andreas Ahland
CTO
Technischer Leiter
Telefon : +49 251 6183-196
Telefax : +49 251 6183-197
Von: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Im Auftrag von Oli Kah
Gesendet: Donnerstag, 7. April 2016 01:13
An:pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Bill,
thanks for answering so promptly =)
I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?!
The three major problems currently for me are:
- it takes much too long to really establish a call (10 seconds are not normal)
- the sound received by the callee sounds strange, as if the sample rate is wrong
- the callee is NOT heard at all on the caller's side!
Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that?
Thank you so much!
Cheers,
Oliver
Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr
Von: "Bill Gardner" <billg@wavearts.com<mailto:billg@wavearts.com>>
An: "pjsip list" <pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>>
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there.
Bill
On 4/6/2016 4:26 PM, Oli Kah wrote:
Hi Bill, hi everyone,
thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h.
The result is exactly the same.
To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log.
There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear).
1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!!
2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected.
3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice!
The Python code of this mini app is attached to this email once more.
I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (seehttp://www.phoner.de/download_en.htm) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow.
Any ideas?
Thank you.
Cheers,
Oliver
Gesendet: Montag, 04. April 2016 um 20:49 Uhr
Von: "Bill Gardner"<billg@wavearts.com><billg@wavearts.com>
An:pjsip@lists.pjsip.org
Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
Hi Oliver,
I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems.
Regards,
Bill
On 4/4/2016 2:37 PM, Oli Kah wrote:
Hmmm, no one?
Is there some sort of forum somewhere where to post things like these??
Thank you :)
Cheers,
Oli
Gesendet: Freitag, 01. April 2016 um 21:38 Uhr
Von: "Oli Kah"<mj_fn@web.de><mj_fn@web.de>
An:pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
Betreff: Sound issues: strange samplerates?
Hi there,
I am new to this list and want to say "Hello" to everyone listening :)
My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible!
What I did:
I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good.
These values were used for my config_site.h:
#define PJMEDIA_CONF_USE_SWITCH_BOARD 1
#define PJMEDIA_SOUND_BUFFER_COUNT 8
#define PJMEDIA_SND_DEFAULT_REC_LATENCY 50
#define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50
The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless!
During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream.
The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box").
The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long?
Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad.
The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge!
What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success.
Thank you!
Cheers,
Oliver
_______________________________________________
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________ Visit our blog:http://blog.pjsip.org pjsip mailing listpjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
__________________________________________________________________________________
SELECTRIC Nachrichten-Systeme GmbH
Haferlandweg 18
48155 Muenster
Internet:
www.selectric.de
---
Sitz der Gesellschaft (Company Premises): Muenster | Registergericht (Register Court): Amtsgericht Muenster | HRB-Nr. 1264
Geschaeftsfuehrer (General Managers): Michael Heussner, Hendrik Pieper
_______________________________________________
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
--
1024D/A9F85A52 2000-01-18 Alain Totouom<totouom@gmx.de>
PGP Fingerprint DA18 0DF2 FBD2 5F67 0656 452D E3A2 7531 A9F8 5A52
_______________________________________________ Visit our blog:
http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________
Visit our blog:http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
_______________________________________________ Visit our blog:
http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
There are config options for each code, so you can get rid of them that
way (look in the code for each codec to see the option), but you'll end
up with both PCMU and PCMA. That's probably OK.
Bill
On 4/7/2016 11:27 AM, Oli Kah wrote:
> Hi Bill,
> thanks for your support :)
> Yes, it seems like the SIP support of my router produced these issues
> and I agree also to remove all codecs that lead to this behavior.
> However I am not working with the command line version of pjsip but
> only using the Python bindings.
> So I have to find a way to eliminate the "unwanted" codecs when
> compiling the Python library somehow.
> Thanks again!
> Cheers,
> Oliver
> *Gesendet:* Donnerstag, 07. April 2016 um 17:19 Uhr
> *Von:* "Bill Gardner" <billg@wavearts.com>
> *An:* pjsip@lists.pjsip.org
> *Betreff:* Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> I agree with the other responders. Disable everything except G711
> (PCMU) using the --dis-codec option and it will probably work better.
> You may also want to wait for the REGISTER to complete before starting
> the INVITE.
>
> Bill
> On 4/7/2016 10:41 AM, Oli Kah wrote:
>
> Hi Alain,
> thank you for answering :) Yes it seems that my VOIP router
> performs things it should not do.
> Currently I am looking for ways to ONLY include one specific
> codec: G711a.
> But I don't know which defines should be set in config_site.h to
> achieve this.
> Cheers,
> Oliver
> *Gesendet:* Donnerstag, 07. April 2016 um 12:00 Uhr
> *Von:* "Alain Totouom" <alain.totouom@gmx.de>
> *An:* "pjsip list" <pjsip@lists.pjsip.org>
> *Betreff:* Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> the callee (Fr!tzBox) did alter the supported codec list
> (offer/answer) between his provisional (183/INVITE/cseq=6911) and
> his final response (200/INVITE/cseq=6911) and did decline the
> callers request (UPDATE/cseq=6912) to use the previously
> negotiated one (G722).
> A more robust solution in a case like this, would i.E. imply
> caching the answer-codec-list and updating the offer-codec-list
> for subsequent calls to the same endpoint…
>
>
> 00:58:12.143 pjsua_core.c .......TX 1276 bytes Request msg
> INVITE/cseq=6911 (tdta003BBA90) to UDP 192.168.1.1:5060:
> [..]
> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
> [..]
>
> 00:58:12.347 pjsua_core.c .RX 755 bytes Response msg
> 183/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
> [..]
> m=audio 7086 RTP/AVP 9 0 8 96
> [..]
>
> 00:58:12.347 pjsua_media.c ......Audio updated, stream #0: G722
> (sendrecv)
> 00:58:34.052 pjsua_core.c .RX 1001 bytes Response msg
> 200/INVITE/cseq=6911 (rdata003392C4) from UDP 192.168.1.1:5060:
> [..]
> m=audio 7086 RTP/AVP 0 8 96
> [..]
>
> 00:58:34.104 pjsua_core.c ....TX 815 bytes Request msg
> UPDATE/cseq=6912 (tdta003D31C8) to UDP 192.168.1.1:5060:
> [..]
> m=audio 4000 RTP/AVP 9 96
> [..]
>
> 00:58:34.155 pjsua_core.c !.RX 353 bytes Response msg
> 488/UPDATE/cseq=6912 (rdata003392C4) from UDP 192.168.1.1:5060:
>
> Cheers,
> Alain
> On 07/04/16 11:14, Andreas Ahland wrote:
>
> HI Oliver,
>
> I remember we had issues with the FritzBox as well as ours did not send the media type which was negotiated in advance. This was clearly a fault in the FritxBox. We got our setup working by using G711a only, i.E. disable G722 and G711u.
>
> Mit freundlichen Grüßen
>
> Dr.-Ing.
> Andreas Ahland
> CTO
> Technischer Leiter
> Telefon : +49 251 6183-196
> Telefax : +49 251 6183-197
>
>
>
>
> Von: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Im Auftrag von Oli Kah
> Gesendet: Donnerstag, 7. April 2016 01:13
> An:pjsip@lists.pjsip.org
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
>
> Hi Bill,
>
> thanks for answering so promptly =)
> I have created the log and yes, there seem to be lots of suspicious lines like "strm003CAB04 Bad RTP pt 0 (expecting 9)" but the rest of the log does not tell me that much and looks mostly as expected?!
>
> The three major problems currently for me are:
> - it takes much too long to really establish a call (10 seconds are not normal)
> - the sound received by the callee sounds strange, as if the sample rate is wrong
> - the callee is NOT heard at all on the caller's side!
>
> Do you see what issues are there in the log that might explain this behavior? And what needs to be done to fix that?
>
> Thank you so much!
>
> Cheers,
> Oliver
>
> Gesendet: Mittwoch, 06. April 2016 um 22:52 Uhr
> Von: "Bill Gardner" <billg@wavearts.com<mailto:billg@wavearts.com>>
> An: "pjsip list" <pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>>
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> Please generate a pjsip logfile (level 4 should suffice) and send, there may be clues in there.
>
> Bill
>
> On 4/6/2016 4:26 PM, Oli Kah wrote:
> Hi Bill, hi everyone,
>
> thanks for answering but I can confirm that the config_site.h settings are NOT the problem. I have now recompiled the lib with nothing in config_site.h.
> The result is exactly the same.
>
> To illustrate what the called person hears I have attached an mp4 which also shows the progress of the Python app in form of PyCharms debug log.
>
> There are important issues that can be seen + heard. Note that I am saying "1-2-3" from the beginning(!) of the video until the very end without ever stopping. Only the receiving side can be heard voice-wise (or not as you will see/hear).
>
> 1) When picking up the phone the sound CANNOT be heard on the receiving side although I am continously saying "1-2-3". It takes roughly 10(!!) seconds before the other side can hear me at all!!
> 2) It also takes 10 seconds after pickup before the call confirmation phase is reached. This is extremly slow and totally unexpected.
> 3) When the other side finally can hear me I sound like a monster... the sample rate seems to be off - I can't otherwise explain the strange sounding voice!
>
> The Python code of this mini app is attached to this email once more.
>
> I don't think I am doing anything exotic. It does not work as expected though. If I do the same with an app like Phoner (seehttp://www.phoner.de/download_en.htm) none of these problems occur within the same environment and using the same accounts and phone numbers. So it has to do with pjsip lib somehow.
>
> Any ideas?
>
> Thank you.
>
> Cheers,
> Oliver
>
> Gesendet: Montag, 04. April 2016 um 20:49 Uhr
> Von: "Bill Gardner"<billg@wavearts.com><billg@wavearts.com>
> An:pjsip@lists.pjsip.org
> Betreff: Re: [pjsip] Fw: Sound issues: strange samplerates?
> Hi Oliver,
>
> I think you should try a completely default configuration, i.e. use an empty config_site.h file. Your config_site.h params may be causing problems.
>
> Regards,
>
> Bill
>
> On 4/4/2016 2:37 PM, Oli Kah wrote:
> Hmmm, no one?
>
> Is there some sort of forum somewhere where to post things like these??
>
> Thank you :)
>
> Cheers,
> Oli
>
> Gesendet: Freitag, 01. April 2016 um 21:38 Uhr
> Von: "Oli Kah"<mj_fn@web.de><mj_fn@web.de>
> An:pjsip@lists.pjsip.org<mailto:pjsip@lists.pjsip.org>
> Betreff: Sound issues: strange samplerates?
> Hi there,
>
>
> I am new to this list and want to say "Hello" to everyone listening :)
>
> My issue using pjsib is rather strange. When calling someone (I tested it using two of my own telephone numbers) I get normal audio first during early call stages and then when the call is finally confirmed the audio rate suddenly is half or so. The voice then sounds monsterish and the remote site is no longer audible via speakers. This happens every time - reproducible!
>
> What I did:
>
> I have successfully compiled pjsua (release build) for Python using Visual Studio 2015 Community linking pjsua to Python 2.6 lib, 32bit running everything on Windows 8.1 Pro 64bit. So far so good.
> These values were used for my config_site.h:
>
> #define PJMEDIA_CONF_USE_SWITCH_BOARD 1
> #define PJMEDIA_SOUND_BUFFER_COUNT 8
> #define PJMEDIA_SND_DEFAULT_REC_LATENCY 50
> #define PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50
>
> The attached python code shall define a simple SIPPhone that will be used in my larger scale application. It's unfinished but I ran into these unresolvable audio problems and hope that you might help me. In its current form the SIPPhone is totally useless!
>
> During testing I have been using the onboard audio card of my mainboard with some Bose speakers and a Samson UB-1 USB microphone which is connected to the Windows-PC using an USB 2.0 port. For the "remote" side I use a dedicated VOIP telephone from Grandstream.
>
> The test code in my attached file is then run using Pycharm + Python 2.6 runtimes. For testing you must replace the numbers + credientials by valid values on your side. Also note that you might have to specify another domain (mine here is "fritz.box").
>
> The called Grandstream rings. Taking up the phone then starts the call confirmation which is going unexpectedly slow. It takes quite some time (between roughly 3-10 seconds) before the "confirmed" status is reached although the audio starts working before that. The scenario is running on my LAN where the SIP server (a FritzBox) is also running here. So quite strange why this takes so long?
> Right after phone pickup the audio can't be heard at all (me constantly talking after phone pickup!) - in both directions! Then after a short time (1-4 sec) the voice can be heard normally and understandably as expected in both directions. But this only works until the "confirmed" status is reached for the call. When it is reached there is no longer any sound coming from the PC speaker and the voice heard on the Grandstream is monsterish (half sample rate?!). The bidirectional audio is lost and the remaining audio is really bad.
>
> The observed behavior happens every time. Just the timing is different and the time from starting the program to hearing the monsterish voice is between 3 and 10 seconds. It is also rather strange that this time span can be so huge!
>
>
> What happens here?! In its current form the SIPPhone is pretty unusable. I hope you have ideas how to fix that :) I have experimented with many of the MediaConfig parameters with no success.
>
>
> Thank you!
>
> Cheers,
> Oliver
>
>
>
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